DE69029658T3 - Method and device for compressing a communication signal - Google Patents

Description

The The present invention relates to communication systems, and more particularly the compression of communication signals increased by one capacity of the communication system.

BACKGROUND OF THE INVENTION

telecommunications systems are well known in the art. From the initial work of Samuel Morse, US Patent No. 1,647 (1840) and Alexander Graham Bell, US Patent 174,465 (1876) has around the globe and beyond a whole telecommunications industry developed.

Either the coding as well as the timing has gone through the historical evolution played a key role in communication systems. For example Messages were written in Morse code before the invention of the phone coded and corresponding electronic pulses transmitted the coded messages via telegraph lines, whereupon they were received and decoded. After that could through the reversal of events an answer delivered become. Manual encoding and decoding of messages prevented the direct real-time communications between two persons.

With the advent of the phone was through the electronic coding of speech patterns in communication signals, their signals via wires between transfer two phones that enabled real-time communication. The speed the electronic communication signal, the speed the sound far exceeded allowed real-time communication between individuals over significant Distances, without a noticeable time delay.

today Communication signals are not limited to wires, but are also through a microwave, radio wave and light guide worn. These advantages have allowed global real-time telecommunications. Furthermore the real-time communications service is expected by the consumer.

Different as in connection with ordinary hard-wired telephone systems, where a single telephone communication signal on a pair of wires guided A time-multiplex technique was used to increase the capacity of the various carrier media to improve. For example, you can many communication signals are multiplexed together and over one led single light guide become. Accordingly, a single fiber optic cable can be a hundred Replace pair wire cables and even provide greater signal transmission performance.

the same Principle was with consideration used on cellular telephone systems. Radiotelephone systems for both stationary and Mobile uses are well known in the art. For example allow cellular phone system in remote rural areas where the installation and maintenance of conventional Telephone wiring is prohibitive, the transmission between base and various subscriber stations to the telephone service to enable. The proliferation of mobile radiotelephone systems in the form of cellular car phones, which are widely available increases more and more.

Radiotelephone systems use a group of selected instead of wire cables Radio frequencies for transmission of communication signals. A typical stationary radiotelephone system can Use 13 pairs of selected frequencies or channels over the the communication signals between subscriber stations and a shared base station and to be received.

As a result the fact that for the radiotelephone use purpose only a limited frequency band allowed, the time-division multiplexing technique was used to capacity of radio telecommunications systems to improve. For example, U.S. Patent No. 4,675,863 a stationary one Radiotelephone system that uses 26 channel pairs, each of which can carry up to four communication signals at once.

Different as for Fiber-optic communications transmissions, the communication signals in the gigahertz range at their and from their destination are carrier radio frequencies (Channels) in their capacity much stronger limited.

Around the capacity of radio channels have been improving voice signal compression schemes used. A procedure that proved successful has residual excitation LPC coding (Residual Excited Linear Predictive Coding - RELP), as disclosed, for example, in WO 86/02726. RELP allowed the compression of a 64 kilobits per second voice communication signal into a 14.6 kilobit-per-second coded signal over the Transmit radio channel becomes. The 14.6 kilobits per second is decoded when it is received to reconstruct a 64 kilobit per second signal that is close to has no noticeable loss in signal quality.

Underlying the mechanism of the RELP is a recursive coding and decoding formula derived from harmonic oscillations depends on the human voice, which in turn provides statistically predictable wave patterns. However, unlike voice transmissions, data communication signals such as modem and facsimile (fax) signals do not exhibit the harmonic characteristics characteristic of voice signals. Accordingly, the RELP signal compression scheme used for voice signals is not quite suitable for fax and modem communication signals. It would be desirable to provide a more suitable encoding compression system for data signals.

Eurocon '86, seventh European conference of the Electrical Engineering, Paris, 21-23 April 1986, pages 484-491, F. Jondral et al the application of digital signal processing and pattern recognition Methods to the automatic classification of high frequency signal "describes a Radio monitoring device the ability has to classify high frequency signals, in turn, on the Learning natural signal sets based. For this purpose, both linear and polynomial Classifier examined.

One Method and system according to the introduction of the respective claims 1 and 11 will appear in "Data Compression of Voiceband Modem Signals ", D. Lin et al., 40th IEEE Vehicular Technology Conference, May 6-9 1990 described.

A Object of the present invention is a radiotelephone system to provide that for the Users regardless of the telecommunications of voice, fax or data is transparent. It is a further object of the invention to provide improved data signal compression method.

The above object and other objects, which will be apparent hereinafter will be through a method and system for transmission of communication signals as defined in the appended claims.

The System is characterized by its identification of the type the communication signal, such as between conversation, fax or modem, as well as the use of different compression methods according to the type of communication signal.

BRIEF DESCRIPTION OF THE DRAWINGS

1 Fig. 10 is a schematic diagram of a radio telecommunications system that may use the improved data compression processing in accordance with the methods of the present invention;

2 Figure 4 is a diagrammatic illustration of the data compression and decoding of communication signals in accordance with the teachings of the present invention;

3 Fig. 12 is a graphical representation of the frequency range of a two-band modem communication signal;

4 Fig. 12 is a graphical representation of the frequency range of a typical facsimile communication signal;

5 Figure 4 is a diagrammatic illustration of the frame structure used in the transmission of a compressed fax signal in accordance with the teachings of the present invention;

6 Fig. 12 is a diagrammatic representation of the frame structure used in the transmission of a compressed modem signal in accordance with the teachings of the present invention;

7 Figure 4 is a diagrammatic representation of the implementation of the improved coding system in a radio telecommunications system.

DETAILED DESCRIPTION THE CURRENT PREFERRED EMBODIMENT

On 1 Referring to Fig. 12, schematically become a base station 11 and a plurality of subscriber stations 10 of a radiotelephone system network. The base station 11 is designed to transmit and receive radio waves at selected frequencies with a number of subscriber stations 10 communicate at the same time. The base station also has an interface with trunk lines 12 a telephone company (TELCO = telephone company). The subscriber units 10 may be stationary to provide, for example, remote area telephone service where the construction of telephone lines is physically impractical and / or cost prohibitive. On the other hand, the subscriber stations 10 be mobile units such as car phones.

One typical system may have 26 predetermined channels in the 450 megahertz spectral range use. The number of channels is generally due to the allocation by the government of chosen Areas of the radio spectrum for Radiotelephone systems limited. For example, the Federal Telecommunications Authority (Federal Communication Commission FCC) in the US in this context certain regulations.

Base station-subscriber station communication is generally over Pairs of frequency channels performed within the specified spectral band. Preferably, the base station transmits Signals on the lower of the two frequencies in each pair and receives signals from the subscriber station to the higher of the two frequencies of each Pair. In a system where 26 frequency channels are available, the base station is built to over 13 different channel pairs to transmit signals simultaneously and to receive.

Around the ability of such a radiotelephone system has been time multiplexed by Communication signals used. For example, in the radiotelephone communications system of the U.S. Patent No. 4,575,863 - the on June 23, 1987 entitled "SUBSCRIBER RF TELEPHONE SYSTEM FOR PROVIDING MULTIPLE SPEECH AND / OR DATA SIGNALS SMULTANEOUSLY OVER EITHER A SINGLE OR A PLURALITY OF RF CHANNELS (Wilson et al.) "Was granted - a telephone system discloses that allows up to four communication signals via transmit a single pair of radio channels become.

Corresponding Each channel pair is divided into four time slots, so that the base station via a single channel pair simultaneously with four different subscriber stations can communicate. This effectively improves the capacity of the radio telephone quad system, so that about the 13 channel pair radio system over 50 telecommunication signals can be transmitted simultaneously. Basically one of the 52 timeslots defined in the 13 channel pairs for performing System additional functions, such as the assignment of channels and time slots for the specific ones Telecommunication signals transmitted with selected subscriber stations be reserved. Such a time-division radio telephone system can be a normal phone service for 500 or more subscriber stations to disposal put.

One such radio telecommunication system that operates using the time-division multiplexing technique is the Ultraphone ^™ system, which is commercially available from the International Mobile Machinery Corporation, the assignee of the present invention.

Around the improved communication skills the radio channel pairs by a time-division multiplex technique to perform compressed the standard communication signals between the participants, to fit into the timeslot created by the time-division multiplexing system granted becomes. For example, with reference to the above-known U.S. Patent 4,675,863 discloses a typical digitized communication signal from 64 kilobits per second to a coded signal of approximately 14.6 Kilobits per second-compressed.

In In practice, a standard analog telecommunication signal will initially be in converted a 64 kilobits per second digital signal. preferably the signal is converted to an 8-bit byte signal, which adds an 8 kilobyte per second digital signal is generated.

in the usual The system referred to above becomes the communication signal processed in steps of 22.5 milliseconds. This leads to a 180 byte sample of the 8 kilobyte per second digital communication signal received in each subsequent frame of the radiotelephone system channel becomes. The time division frame for each pair of channels is built to be 14.6 kilobits per second encoded signal. For This is synonymous with a given 22.5 millisecond frame with 41 8-bit bytes of information per frame. Accordingly must for each Frame the information contained in the 180 byte sample in not more than 41 bytes for transmission coded into one of the time slots of a selected frequency channel become. Furthermore have to the coded 41 bytes when received at the receiving site in the 180 bytes sample for be reconstructed every frame, without that in the communication signal information is distorted or lost.

For voice transmissions it is known a residual excitation LPC coding system (RELP) to encode a 180 byte sample into a 41 byte sample To process the signal in an acceptable equivalent 180 byte sample can be reconstructed. The RELP coding system depends on the use of certain inherent pitch features of a voice signal. On such a coding system is in US Pat. No. 4,675,863 and described in detail in US Pat May 9, 1986 published PCT International Publication No. WO 86/02726, this patent application hereby is incorporated by reference as if fully explained below.

While the RELP coding has been found to be satisfactory for voice signals, it is not fully suitable for the encoding of fax (facsimile) and / or modem communication signals. These signals do not show the harmonic vibration and elevation characteristics of a voice signal. Accordingly, the underlying recursive algorithms on which the RELP is based do not adequately allow the encoding of such signals. Nonetheless, the hardware processors available for the RELP data compression are also suitable for the compression procedure of the present invention. For example, the Texas Instrument Model TMS 32020 digital signal processor is preferred for performing the instant compression method.

To improve the transmission of fax and modem signals in such radiotelephone communications systems, an improved coding method and system has been devised therefor. With reference to 2 An applicant's inventive decoding and coding system for communication signals - in particular fax and data signals - is shown schematically.

The processing of the communication signal from its standard analog form into a digital 64 kilobits per second (8 kilobyte per second) digital signal for processing in the 180-byte sample per frame is standardized within the overall radio telecommunication system for processing all communication signals. When faxes and / or data signals are transmitted, it becomes 8 kilobytes per second of digital signal 20 through a coding processor 22 in an optionally formatted coded signal 21 coded. The coded signal 21 will be in a selected time slot of one of the radio channels 24 of the system, all of which contain a plurality of time slots defined by a time division multiplex technique. The receiving site includes a decoding processor 25 , The coded signal 21 becomes a decoder for processing 25 directed to a communication signal 26 to reconstruct, which is substantially the same as the call signal. Both the transmission site and the receiving site include a compression selection processor 27 . 28 one that coordinates the respective encoding and / or decoding activity, as described below with reference to FIG 7 Basically, each site includes both an encoder 22 and a decoder 25 for duplex station-to-station communication. Moreover, the compression selection processor, encoder and decoder can all be implemented in a single microprocessor such as a Texas Instrument Model TMS 32020 Digital Signal Processor.

The coding processor 22 interpolates the communication signal 20 by a selected factor M 'to increase the size of the sample. The magnified sample is multiplied by the cos (Ωt) and sin (Ωt) into its respective in-phase and quadrature components 30 . 31 divided up. This results in the simultaneous processing of two information bit streams 30 . 31 , Ω is chosen as the approximate center frequency of the frequency domain representation of the signal. The respective mixing by cos (Ωt) and sin (Ωt) shifts the frequency range of the communication signal from Ω to 0 Hz, after mixing, each respective signal 30 . 31 Low-pass filtered to remove the frequency components above a selected level in the frequency domain. This eliminates the distortion and echo frequency components centered on even multiples of Ω.

Each filtered sample 32 . 33 is then decimated by a selected factor M to reduce the sample size for quantization. The decimated signal 34 . 35 for both the in-phase and quadrature components is then quantized into a selected number of levels, with an adaptive pulse code modulator encoder, resulting in quantized signal samples 36 . 37 and a quantization gain component G. The respective quantized signal samples and the amplifier component are then combined with the unique word 40 in the coded signal 21 formatted, which has a selected frame structure. Although separate amplifier components can be generated for the in-phase and quadrature components, a common gain value of the most significant 8 bits is satisfactory for compressing the 8 kilobyte per second signal into a 14.6 kilobits per second encoded signal.

The frame structure conforms to the format requirements of the slots allocated for the communication signals in the time-division multiplexed channels of the radio telephone system. In the preferred system, the frame format is 41 Bytes per frame. The clear word 40 carries information concerning the transmitted signal type, ie voice, fax or modem, and the timing information. In the receiver, the coded frame is processed in accordance with this information.

The decoding processor 25 The receiving station separates the quantized signal samples 36 . 37 and the quantization gain G for the respective in-phase and quadrature components. The decoding of the quantized signals 36 . 37 is then performed in accordance with the quantization amplifier parameter G, resulting in the communication signal samples 42 . 43 which, in terms of information, equals the pre-quantized, decimated samples 34 . 35 are. After that, the signal samples 42 . 43 for both the in-phase and quadrature components are successively interpolated by the factor M.

The interpolated in-phase signal 44 is then mixed again with cos (Ωt) and the interpolated quadrature signal 45 is then mixed again with sin (Ωt). Both signals 46 . 47 then be on the low-pass filtered level for which filtering was performed following the initial mixing of the signals. Every filtered signal 48 . 49 is decimated by the factor M 'and the two signals are added together to form a communication signal 26 to reconstruct that equivalent to the initial 8 kilobyte signal 20 is. Although it is possible to synchronize the encoding and decoding processes, synchronization is not required.

In the preferred embodiment, which is compatible with the system disclosed in U.S. Patent 4,675,863, when the digitized 8 kilobyte per second communication signal is encoded 20 is encoded, the encoding processor 22 a 180 byte sample into a 41 byte frame structure for time division multiplex transmission over a selected radiotelephone system channel.

As in 3 For example, two-band modem communication signals are centered rather narrowly at either 1200 hertz (representing data transmission from the original modem) or 2400 hertz (representing data transmission from the answering modem). Fax communication signals are usually around 1800 hertz in a wider range, such as in 4 illustrated, centered.

Preferably, when a fax signal is transmitted, the signal is interpolated by a factor of three, thereby increasing the frame sample size to 540 samples per frame. The mixing is then performed for the in-phase and quadrature components by cos (1800t) and sin (1800t), respectively. Low pass filtering is performed to eliminate frequencies higher than 1400 Hertz. The respective in-phase and quadrature components 32 . 33 are then decimated by a factor of 10, resulting in a reduction of the sample size to 54 samples for each component of the frame.

The in-phase and quadrature samples 34 . 35 are then quantized in 6 stages, with an adaptive pulse code modulation encoder, resulting in the quantized representation of the samples 36 . 37 and a quantization gain G of 8 bits. The encoded quantized representations of the respective 54-byte samples per frame of each component are encoded into 8-bit bytes, each of which represents a group of three quantized signal samples. Thus, the 54 samples of the respective components remaining after decimation are represented by 18 bytes, each representing a quantized value of three of the 54 samples and the 8-bit gain G.

Accordingly, a total of 36 8-bit bytes D1-D36 and the 8-bit gain G for data transmission by the radio telecommunications system - such as the in 5 shown - formatted into a 41-byte frame. The 16-bit unique word U, a 4-bit error check code C and 12 unused bits X fill the frame to complete the 41 bytes.

After the frame is transmitted and received over the carrier medium, the formatted frame is then decoded by separating the 8-bit gain G and the 18 in-phase and 18-quadrature quantized bytes. The quantized signals 36 . 37 are then respectively decoded in accordance with the quantization gain G, resulting in 54 8-bit samples 42 . 43 which contains the information equivalent to the respective pre-quantized decimated in-phase and quadrature signals.

The respective decoded in-phase and quadrature samples 42 . 43 are then interpolated by a factor of 10 to increase the sample size to 540. The resulting signal 44 the in-phase component is then mixed with cos (1800t). Similarly, the quadrature component becomes 45 mixed with sin (1800t). The samples 46 . 47 are then low-pass filtered to remove the frequency domain components above 1400 Hertz. Thereafter, the resulting in-phase and quadrature components become 48 . 49 decimated by a factor of 3 to reduce the sample size to 180 8-bit samples in the frame. Finally the signals 50 . 51 summed to a communication signal 26 This is essentially equivalent to the initial 8 kilobytes per second signal 20 is.

For the modem transmission are the parameters for the data compression used to be slightly different. As with The fax signals will transmit digital signals every 180 bytes per frame interpolates a factor of 3 to 540 the sample size to enlarge. The Mixing the signal into an in-phase and quadrature signal by cos (Ωt) and sin (Ωt) so performed that Ω, depending on whether the signal is broadcast from the call or answering modem is equal to either 1200 hertz or 2400 hertz.

The low-pass filters is applied with a limit frequency of 700 hertz. The deeper one Level of low-pass filtering for modem signals Compared to fax signals, this is due to the rather narrow bandwidth the frequency range of the modem signal around 1200 and 2400 hertz.

After filtering, the signal is through ei decimated by a factor of 20 to reduce the samples to 27 samples per frame. The respective in-phase quadrature samples are quantized with an adaptive PCM encoder in 32 levels. This results in 27 five-bit quantized representations D1-D54 of the decimated samples for each of the respective in-phase and quadrature components along with an 8-bit quantization gain G. This information, along with the 16-bit unique word U, a four-bit Error Check Code C and 12 Unused Bits X is formatted into a 41 byte frame structure for transmission in the selected time slot of the selected frequency channel pair over which the radio telecommunications are performed. 6 represents the frame structure for this data communication. Note that preferably the unique word U is formatted in the same position regardless of the type of signal.

At the receive end, the received 41-byte frame is separated into the respective 27 five-bit quantized samples for the in-phase and quadrature component and the 8-bit quantization gain G. The encoded five-bit quantized samples are decoded in accordance with the quantization gain G to result in 27 8-bit signal samples corresponding in information to the respective decimated in-phase and quadrature signal components. The decoded samples are interpolated by a factor of 20 to result in 540 samples per frame. Again, these samples are mixed with cos (Ωt) and sin (Ωt) and then low-pass filtered using the same filter level (700 Hz) as in the transmitter. After that, the mixed and filtered decoded signals 48 . 49 decimated by a factor of 3 to result in 180 samples per frame. The two signals 50 . 51 are then summed to the 8 kilobyte per second communication signal 26 to generate that from the information forth the call signal 20 equivalent.

The clear word 40 is used to indicate the processed signal type so that the system uses the appropriate compression method and associated parameters with the particular signal. For example, the unique word will indicate whether the communication signal must be processed as a voice, fax, modem call or modem response signal.

A prior art cellular telecommunication system, such as that described in U.S. Patent No. 4,675,863, may be modified immediately to use the inventive compression method. 7 schematically illustrates the change that the substitution of a compression selection processor 60 (CSP = compression selection processor) and the associated coding / decoding processors (CODECs) 61 . 62 . 63 . 64 for each voice encoder / decoder (CODEC) in the prior art system.

In general, the CSP uses 60 just one of the CODECs 61 - 64 at a given time. Accordingly, all CODECs can be integrated into a single microchip, with the CSP controlling the parameters and encoding method used for each given communication signal. In fact, all processoars can 60 - 64 integrated into a Texas Model TMS 32020 Digital Signal Processor to perform both the coding selection and the appropriate encoding and decoding process.

Preferably uses the radio telecommunication system in the compression processing of Communication signals a desired Voice signal compression method such as RELP as the default state. This is preferred there at the beginning of the fax and modem transmission a standard echo canceling pause sound either at 2225 or 2100 Hz is generated.

The compression selection processor 60 monitors the communication signal 20 that through the voice CODEC 61 is processed to check for an echo canceling pause. This is done to check the first two reflection coefficients of each frame. If these coefficients are within a certain range for a sufficient number of frames, the system switches from voice processing to processing the communication signal in accordance with the data compression approach of the present invention in the fax and modem CODECs 62 - 64 ,

After switching from the voice state, the system initially processes the communication signals with the fax CODEC 62 in accordance with the parameters for the fax signal transmission discussed above. The communication signal 20 is then monitored to detect the presence of the fax signaling. Specifically, detection is performed by utilizing the presence of a 300 bit per second half-duplex FSK (Frequency Shift Keying) signal (using 1650 and 1815 Hz) used for initial handshaking between fax machines.

The FSK signal is detected in the following manner. A second degree LPC analysis is performed on the signal producing 2 reflection coefficients. Each reflection coefficient is compared with the ent averaging coefficients averaged over the previous 3 frames. If the mean value of the coefficients falls within the set of predetermined limits, the facsimile transmission is detected and the processing proceeds in accordance with the facsimile CODEC 62 continued, ie using the compression procedure described above, which uses fax parameters. However, if the FSK signal is not detected within a specified time window, such as 4.725 seconds, the system will begin the appropriate modem CODEC 63 . 64 to use.

Upon detection of the lock tone and the absence of FSK signal acquisition, the originating modem becomes CODEC 63 used. The unique word transmitted in each frame is processed by the receiving site to decode the frame as voice, fax and modem call or modem response data, respectively. When the receiving station detects receipt of a unique word indicating a calling modem signal, the selection processor switches 60 the receiving station to the answering modem CODEC 64 to use for the return signals.

In addition to monitoring the transmission direction communication signal 20 for fax signaling, the processor monitors 60 also the energy in the transmission direction. When the energy from the transmission direction disappears for a predetermined interval, such as 67.5 milliseconds for modem signals and 22.5 milliseconds for facsimile signals, the processor notices this and the system returns to the default state by using the communication signal as a voice signal Voice CODEC 61 is processed. Regardless of whether the FSK signal is detected, the energy in the transmission direction is constantly monitored to determine the end of the fax and / or modem signaling to reset the system to voice signaling.

Even though the present data compression method in conjunction with a specific radio telecommunication system in a preferred embodiment described, it can be immediately related to other systems customized in which parameters, frequencies, the carrier medium, the frame timing and structure changed become. additionally the parameters used in the data compression method were referenced for compatibility with the systems described in U.S. Patent 4,675,863. It is possible, more sentences to formulate parameters that effectively affect the compression of the communication signal in accordance with the disclosed invention, in accordance with the disclosed method in the information corresponding Data signals can be decoded.

Claims (26)

Method for transmitting communication signals ( 20 ) of different types between different bodies ( 10 . 11 ) via a selected carrier medium ( 24 ) of a telecommunication system, wherein the communication signal ( 20 ) is compressed to enable its transmission over the selected carrier medium ( 24 ) and the communication signal ( 20 ) is reconstructed upon receipt, the method comprising the following steps: - converting the communication signal ( 20 ) into two separate components, including: determining the approximate center frequency Ω of the communication signal ( 20 ), - mixing the communication signal ( 20 ) with cos (Ωt) for generating an in-phase signal component ( 30 ), and - mixing the communication signal ( 20 ) with sin (Ωt) for generating a quadrature signal component ( 31 ); and quantizing each separate in-phase and quadrature signal component ( 30 . 31 ), whereby the separated signal components ( 30 . 31 ) into a quantized in-phase and a quantized quadrature signal ( 36 . 37 ) and an associated quantization gain parameter (G) to encode a compressed coded signal (G). 21 ) for transmission over the selected carrier medium ( 24 ) of the communication system, characterized in that the method further comprises the step of inserting a unique word (U) into the compressed coded signal to indicate the type of signal being transmitted. A method according to claim 1, further characterized by: the transmission of the compressed coded signal ( 21 ) over the selected carrier medium ( 24 ) of the communication system and the reception of the compressed coded signal ( 21 ) in a receiving station; the separation of the quantized in-phase and quadrature signals ( 36 . 36 ) and the amplification component (G) of the compressed coded signal ( 21 ), which is transmitted via the selected carrier medium ( 24 ) of the communication system is received; the determination of the approximate center frequency Ω 'of the compressed coded signal ( 21 ) which is decoded; the reconstruction of the quantized in-phase and quadrature signals ( 36 . 37 ) by quantization decoding in accordance with Ver amplification component (G) in order to obtain reconstructed in-phase and quadrature signals ( 42 . 43 ) to create; the mixture of the reconstructed in-phase signal ( 42 ) with cos (Ω't) to produce a mixed reconstructed in-phase signal ( 46 ) to create; the mixture of the reconstructed quadrature signal ( 43 ) with sin (Ω't) to produce a mixed reconstructed quadrature signal ( 47 ) to create; and the summation of the respective mixed reconstructed in-phase and reconstructed quadrature signals ( 46 . 47 ), a decoded decompressed communication signal ( 26 ) to reconstruct. A method according to claim 2, further characterized by filtering all separate in-phase and quadrature signal components ( 30 . 31 ) before quantizing to remove from a frequency range of the component all frequencies above a selected level to form a respective filtered in-phase and quadrature component ( 32 . 33 ) and by formatting the quantized in-phase and quadrature signals ( 36 . 37 ) and the amplification component (G) to the compressed coded signal ( 21 ) for transmission over the selected carrier medium ( 24 ) of the communication system. A method according to claim 3, further characterized by the decimation of the respective filtered in-phase and quadrature components ( 32 . 33 ) is quantized by a selected factor M, before the respective decimated in-phase and quadrature components ( 34 . 35 ) to create. A method according to claim 4, further characterized by interpolating the communication signal ( 20 ) by a selected factor M 'before the communication signal ( 20 ) is mixed with the cos (Ωt) and the sin (Ωt) to form the respective isolated in-phase and quadrature signal components ( 30 . 31 ) to create. A method according to claim 5, further characterized by filtering the respective mixed reconstructed in-phase and quadrature signals ( 46 . 47 ), before summing to remove from a frequency range of each component all frequencies above the selected filtering level to produce the respective filtered reconstructed in-phase and quadrature signals ( 48 . 49 ) to create. A method according to claim 6, further characterized by interpolating the reconstructed in-phase and quadrature signals ( 42 . 43 ) by a selected factor m before mixing. A method according to claim 7, further characterized by the decimation of the respective filtered reconstructed in-phase and quadrature signals ( 48 . 49 ) by a selected factor m 'before summation. A method according to claim 8, further characterized by determining the approximate center frequency Ω of the communication signal ( 20 ), by the determination of the type of communication signal, and by the assignment of predetermined values for processing the communication signal ( 20 ) based on the signal type. A method according to claim 9, further characterized in that: the predetermined value assigned to Ω is 1800 ( 4 ), M is selected to be 10, M 'is selected to be 3, the filtering level is selected to be 1400 Hz, and the respective decimated in-phase and quadrature components ( 34 . 35 ) are quantized in 6 stages, if it is determined that the communication signal ( 20 ) is a fax signal; or the predetermined Ω value 1200 is ( 3 ), M is selected to be 20, M 'is selected to be 3, the filtering level is selected to be 700 Hz, and the respective decimated in-phase and quadrature components ( 34 . 35 ) are quantized in 32 stages, if it is determined that the communication signal ( 20 ) is a call band signal of a two-band modem signal; or the predetermined Ω assigned value 2400 is ( 3 ), M is selected to be 20, M 'is selected to be 3, the filtering level is selected to be 700 Hz, and the respective decimated in-phase and quadrature components ( 34 . 35 ) are quantized in 32 stages, if it is determined that the communication signal ( 20 ) is a response band signal of a two-band modem signal; or the predetermined value Ω 'is 1800 ( 4 ), m is selected to be 10, m 'is selected to be 3, and the filtering level is selected to be 1400 Hz when it is determined that the compressed coded signal (Fig. 21 ), the communication signal ( 20 ), is a fax signal and is received at a receiving station; or the predetermined value Ω 'is 1200 ( 3 ), m is selected to be 20, m 'is selected to be 3, and the filtering level is selected to be 700 Hz, when it is determined that the compressed coded signal (Fig. 21 ), the communication signal ( 20 ), a call band signal of a two-band modem signal is received at a receiving station; or the value 2400 assigned to the predetermined Ω 'is ( 3 ), m is selected to be 20, m 'is selected to be 3, and the filtering level is selected to be 700 Hz, when it is determined that the compressed coded signal (Fig. 21 ), the communication signal ( 20 ), a response band signal of a two-band modem signal is received at a receiving site. Telecommunication system for the transmission of communication signals ( 20 ) of different types between different bodies ( 10 . 11 ) via a selected carrier medium ( 24 ), the communication signal ( 20 ) is compressed to enable its transmission over the selected carrier medium ( 24 ) and the communication signal ( 21 ) is reconstructed after receiving; the telecommunication system comprising a signal compression coder ( 22 ), wherein the encoder ( 22 ) comprises: means for converting a communication signal into two separate components, including: a means ( 27 ) for determining the approximate center frequency Ω of the communication signal ( 20 ), - means for mixing the communication signal ( 20 ) with cos (Ωt) to obtain an in-phase signal component ( 30 ), and - a means for mixing the communication signal ( 20 ) with sin (Ωt) to form a quadrature signal component ( 31 ) to create; and - means for quantizing each separate in-phase and quadrature signal component ( 30 . 31 ), whereby the separated signal components ( 30 . 31 ) into a quantized in-phase and a quantized quadrature signal ( 36 . 37 ) and an associated quantization gain parameter (G) to encode a compressed coded signal (G). 21 ) for transmission over the selected carrier medium ( 24 ) of the communication system, characterized in that the encoder further comprises means for inserting a unique word (U) into the compressed coded signal (U) 21 ) for indicating the type of transmitted signal. A system according to claim 11, further characterized in that the encoder ( 22 ) means: means for filtering each separated in-phase and quadrature signal component ( 30 . 31 ) prior to quantization to remove from a frequency range of each component all frequencies above a selected level to thereby produce a respective filtered in-phase and quadrature component ( 32 . 33 ) to create; and formatting means for multiplexing the quantized in-phase and quadrature signals ( 36 . 37 ) and the amplification component (G), thereby expressing the compressed coded signal (G) 21 ) for transmission over the selected carrier medium ( 24 ) of the communication system. A system according to claim 12, further characterized in that the encoder ( 22 ) means for decimating the respective filtered in-phase and quadrature components ( 32 . 33 ) by a selected factor M before quantization. A system according to claim 13, further characterized in that the encoder ( 22 ) means for interpolating the communication signal ( 20 ) by a selected factor M 'before the communication signal ( 20 ) is mixed with the cos (Ωt) and the sin (Ωt) to form separate in-phase and quadrature signal components ( 30 . 31 ) to create. A system according to claim 14, characterized in that the means ( 27 ) for determining the approximate center frequency Ω of the communication signal ( 20 ) determines the type of communication signal and assigns Ω, M, M 'preselected values, the filtering level and the number of quantization stages being based on the signal type. A system according to one or more of claims 11-15, further characterized in that it includes: a signal decompression decoder ( 25 ) comprising: format decoding means for separating the quantized in-phase and quadrature signals ( 36 . 37 ) and the amplification component (G) of the compressed coded signal ( 21 ), which is transmitted via the selected carrier medium ( 24 ) of the communication system is received; a means ( 28 ) for determining the approximate center frequency Ω 'of the compressed coded signal ( 21 ) which is decoded; a quantization decoding means for the reconstruction of the in-phase and quadrature signals ( 36 . 37 ) in accordance with the gain component (G) to produce reconstructed in-phase and quadrature signals ( 42 . 43 ) to create; a means for mixing the reconstructed in-phase signal ( 42 ) with cos (Ω't) to produce a mixed reconstructed in-phase signal ( 46 ) to create; a means for mixing the reconstructed quadrature signal ( 43 ) with sin (Ω't) to produce a mixed reconstructed quadrature signal ( 47 ) to create; and means for summing the respective mixed reconstructed in-phase and quadrature signals ( 46 . 47 ), a decoded decompressed communication signal ( 26 ) to reconstruct. A system according to claim 16, further characterized by a first communication station ( 10 ), the compression encoder ( 22 ), and by a second communication station ( 11 ) for receiving the compressed coded signal ( 21 ) from the first station ( 10 ), the signal decompression decoder ( 25 ) having. A system according to claim 16, further characterized by a plurality of communication stations ( 10 . 11 ), each comprising at least one of the signal compression coders ( 22 ) and at least one of the signal decompression decoders ( 25 ) having. A system according to claim 18, further characterized in that at least one of the communication stations ( 10 ) comprises: a plurality of the signal compression encoders ( 22 ) and a plurality of signal decompression decoders ( 25 ) for encoding and decoding a selected signal type; means for selecting one of the plurality of encoders ( 22 ) for processing a signal to be transmitted ( 20 ) in accordance with the signal type; and means for selecting one of the plurality of decoders ( 25 ) for processing a received signal ( 21 ) in accordance with the signal type. A system according to any one of claims 16 to 19, characterized in that the means ( 28 ) for determining the approximate center frequency Ω 'of the compressed coded signal ( 21 ) determines the type of communication signal and assigns Ω 'a predetermined value based on the signal type. A system according to any one of claims 16 to 19, characterized in that: each signal compression encoder ( 22 ) further comprises means for filtering each of the separate in-phase and quadrature signal components ( 30 . 31 ) before quantizing to remove from the frequency range of each component all frequencies above a selected level; and in that each signal decompression decoder ( 25 ) further comprises means for filtering the respective mixed reconstructed in-phase and quadrature signals ( 46 . 47 ) before summing to remove from the frequency range of each component all frequencies above a selected filtering level to produce respective filtered reconstructed in-phase and quadrature signals ( 48 . 49 ) to create. A system according to claim 21, characterized in that: each signal compression encoder ( 22 ) further comprises means for decimating the respective filtered in-phase and quadrature components ( 32 . 33 ) by a selected factor M before being quantized to form a respective decimated in-phase and quadrature component ( 34 . 35 ) to create; and in that each signal decompression decoder ( 25 ) further comprises means for interpolating the reconstructed in-phase and quadrature signals ( 42 . 43 ) by a selected factor m before mixing. A system according to claim 22, characterized in that: each signal compression coder ( 22 ) further comprises means for interpolating the communication signal ( 20 ) by a selected factor M 'before the communication signal ( 20 ) is mixed with the cos (Ωt) and the sin (Ωt) to form the respective isolated in-phase and quadrature signal components ( 30 . 31 ) to create; and each signal decompression decoder ( 25 ) further comprises means for decimating the respective filtered reconstructed in-phase and quadrature signals ( 48 . 49 ) by a selected factor m 'before summing. A system according to claim 23, characterized in that: the encoder ( 22 ) a means ( 27 ) for determining the type of communication signal and to each encoder ( 22 ) assigns predetermined values for Ω ', M, M', wherein the filtering level and the number of quantization steps are based on the signal type ( 20 ); and the decoder ( 25 ) a means ( 28 ) for determining the type of communication signal and assigning to each decoder predetermined values for Ω ', m, m', the filtering level and the number of quantization levels being based on the type of signal. A system according to claim 24, characterized in that: the predetermined value assigned to Ω is 1800 ( 4 ), M is selected to be 10, M 'is selected to be 3, the filtering level is selected to be 1400 Hz, and the respective decimated in-phase and quadrature components ( 34 . 35 ) are quantized in 6 stages, if it is determined that the communication signal ( 20 ) is a fax signal; or the predetermined Ω value 1200 is ( 3 ), M is selected to be 20, M 'is selected to be 3, the filtering level is selected to be 700 Hz, and the respective decimated in-phase and quadrature components ( 34 . 35 ) are quantized in 32 stages, if it is determined that the communication signal ( 20 ) is a call band signal of a two-band modem signal; or the predetermined value Ω is 2400 ( 3 ), M is selected to be 20, M 'is selected to be 3, the filtering level is selected to be 700 Hz, and the respective decimated in-phase and quadrature components ( 34 . 35 ) are quantized in 32 stages, if it is determined that the communication signal ( 20 ) is a response band signal of a two-band modem signal; or the value Ω assigned to the predetermined Ω 'is 1800 ( 4 ), m is selected to be 10, m 'is selected to be 3, and the filtering level is selected to be 1400 Hz when it is determined that the compressed coded signal (Fig. 21 ), the communication signal ( 20 ), is a fax signal and is received at a receiving station; or the predetermined value Ω 'is 1200 ( 3 ), m is selected to be 20, m 'is selected to be 3, and the filtering level is selected to be 700 Hz, when it is determined that the compressed coded signal (Fig. 21 ), the communication signal ( 20 ), a call band signal of a two-band modem signal is received at a receiving station; or the predetermined value Ω 'is 2400 ( 3 ), m is selected to be 20, m 'is selected to be 3, and the filtering level is selected to be 700 Hz, when it is determined that the compressed coded signal (Fig. 21 ), the communication signal ( 20 ), a response band signal of a two-band modem signal is received at a receiving site. A system according to claim 25, characterized in that the means ( 27 ) for determining the approximate center frequency Ω of the communication signal ( 20 ) determines the type of communication signal.